Bugfix that fixes the integration issue that cause WebRTC AEC mobile to fail

webrtc/voice_engine/output_mixer.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 531 matching lines @@
     // --- Measure audio level (0-9) for the combined signal
     _audioLevel.ComputeLevel(_audioFrame);
 
     return 0;
 }
 
 // ----------------------------------------------------------------------------
 //                             Private methods
 // ----------------------------------------------------------------------------
 
 void OutputMixer::APMProcessReverseStream() {

[aluebs-webrtc, 2015/12/15 19:16:08]

APMAnalyzeReverseStream?

[the sun, 2015/12/15 19:37:14]

If the name is a problem I'd suggest making this code inline in the one place
where it is being called.

[peah-webrtc, 2015/12/15 21:29:45]

I decided to keep it since I expected that this CL would be followed up with
another fix that allowed the ProcessReverseStream call to be reintroduced. I
will change according to the inline suggestion instead though. Please let me
know if you are ok with that!

[peah-webrtc, 2015/12/15 21:29:45]

Done.

[aluebs-webrtc, 2015/12/15 21:40:34]

Sounds good to me.

-  if (_audioProcessingModulePtr->ProcessReverseStream(&_audioFrame) != 0) {
-    WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
-                 "AudioProcessingModule::ProcessReverseStream() => error");
+  // Convert from mixing to AudioProcessing sample rate, similarly to how it
+  // is done on the send side. Downmix to mono.
+  AudioFrame frame;
+  frame.num_channels_ = 1;
+  frame.sample_rate_hz_ = _audioProcessingModulePtr->input_sample_rate_hz();
+  RemixAndResample(_audioFrame, &audioproc_resampler_, &frame);

[the sun, 2015/12/15 19:37:14]

What are the expected performance effects of this?

[peah-webrtc, 2015/12/15 21:29:45]

In what sense do you mean? There is a computational cost for the resampling and
how big that is I cannot tell. But note that before Aug 14, this resampling was
always done so it should not be anything that the platforms where it is used
cannot handle.

The resampling is needed to at all be able to run AECM but on a longer term
basis we should perhaps consider to do it in the band-splitting mechanism
instead.

+
+  if (_audioProcessingModulePtr->AnalyzeReverseStream(&frame) != 0) {
+    WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+                 "AudioProcessingModule::AnalyzeReverseStream() => error");
+    assert(false);

[hlundin-webrtc, 2015/12/15 16:07:05]

DCHECK

[peah-webrtc, 2015/12/15 21:29:45]

Done.

   }
 }
 
 int
 OutputMixer::InsertInbandDtmfTone()
 {
     uint16_t sampleRate(0);
     _dtmfGenerator.GetSampleRate(sampleRate);
     if (sampleRate != _audioFrame.sample_rate_hz_)
     {
@@ skipping 29 matching lines @@
             _audioFrame.data_[2 * i + 1] = 0;
         }
     }
     assert(_audioFrame.samples_per_channel_ == toneSamples);
 
     return 0;
 }
 
 }  // namespace voe
 }  // namespace webrtc